similar to: SIP port 5060 closed - how do I open it?

Can anyone tell me why my inbound calls keep getting rejected with 401? Here's the debug information: <--- SIP read from UDP:147.135.32.221:5060 ---> INVITE sip:6087294351 at 216.26.109.22:5060 SIP/2.0 Call-ID: 31007e-31 at 147.135.32.221 CSeq: 1 INVITE From: "Wi M"<sip:4144038968 at 147.135.32.221;user=phone>;tag=9bbc To: "Gregory Malsack"<sip:s at

Hi everyone, having a issue with asterisk and my new Voip providers service. Iv set up many asterisk systems before but never seen this and have tried to fix this with no luck.. I have used this exact same sort of setup for 5 other providers and never had this issue, If i replace the trunk login details with my works voip account and set it to IAX then it works perfect, Just not the new

Hi, I've a few working asterisk servers, all seeing the same symptom, but they are all based on the same configs. A SIP inbound INVITE message is coming in to an extension (not a peer) eg 555 at ourserver.com A tcpdump clearly shows the INVITE coming in, but asterisk seems to be ignoring it (theres no reply outbound packet). All the source/dest IPs and ports look good. A

Hello, We have 2 SIP trunks from Bandwidth.com and if both are in use and someone tries to dial out, they cause another call to get one-way audio (the caller hears us, we cannot hear them). This happens 100% of the time and Bandwidth.com doesn't offer any support. I don't see any setting that tells Asterisk that there are 2 channels available from Bandwidth.com's IP. I'm

Hi all, I am using an asterisk as a client to connect to another asterisk server by registering with the register string. Registration is done without any hassel, but after sometime my asterisk loses the registration with the server and the server starts displaying the following msgs repeatedly: [Aug 9 06:37:59] NOTICE[8380]: chan_sip.c:8151 check_auth: Correct auth, but based on stale nonce

We have multiple entries like the one below in our users.conf file... where the username. Contact and secret changes for different customers and we register on their behalf to the Service Provider. For the trunk below: when the call is placed out, Asterisk (1.4.18) sends the username of "abc.com" in the MD5 Auth .....which obviously does not match the trunk setup for this Customer with

Hello All, Some new security stuff is going on I suppose in 1.8 that I am not familiar with and would appreciate your help In a scenario such as the following: Internet --> SBC --> Asterisk upon trying to register an endpoint, the following is being observed on the Asterisk Console. Have Googled this but haven't come up with anything that helped much. [Mar 10 11:53:59] ERROR[21272]:

I got no resonses to this and some funny bounces so I'm trying again. First of all Merry Christmas. Second, my first problem with my provider not staying registered with our server was my fault. We moved our server room and I restarted the test system and the production system causing them to ping-pong back and forth registering with our provider causing random problems, they are both

Hi, I am trying to setup a simple home voip service w/ * I have compiled and installed the svn source as a first step I am trying to configure SIP for inside my network. I have a handful of softphones and a few hardphones that I want to all be able to call each other I have configured users.conf with a single softphone(kphone) and have tried calling itself (ext 6000) and the demo from the

Hello. I don't know if this is a problem, but I was expecting a different behavior. Users, have to dial "0" to get an external line, and afterwords the number they want to dial (exe 12345). The thing is: 1-If user dial "012345" there is an error and the call isn't made and the error is "handle_request_invite: Call from 'XXX' to extension '012345'

Hi all, I'm not getting any sidetone on my Snom 370. I searched the web and the snom wiki, but I don't see any place to enable/adjust it. Callers say I sound great on the other end, but I don't hear myself so it is a little off-putting. Any suggestions would be appreciated. On a related note, some times (maybe 1 out of 10 calls) I get the side tone, but its delayed by a second or

Hi All, I have done some research on Asterisk and I would like to try it in my office. Here's what I'm looking at for my system: AsteriskNow running one of two ways: 1) As a virtual machine on a VMWare server (Eight core Xeon server with 4GB ram) 2) on a P4 2.4Ghz with 768mb RAM I'm looking at 4-5 phones in the office. I was going to go with Grandstream or Polycom phones

I need some extensions logic assistance, I'm trying to dial out one of multiple SIP trunks, in sequence. I need to detect a busy SIP trunk(I only allow 1 call per trunk) and roll over to a second or third depending on that busy status Here's what I've got for a macro thusfar, but it's not working(fails if the 1st trunk is busy) extensions.conf: [globals] trunk_1 => SIP/trunk1

Hey I want to generate a discrete uniform distribution as follows: For example: I want to get 278734 records each with a numbers between 7-10. And the sum of numbers in 278734 records to be equal to 2253712. Once this is done, I want to get that printed to an excel file such that Record Value 1 9 2 7 so on 278734 8 The

Thank you. While I'm currently in the process of migrating to AlmaLinux 8 via an in-place upgrade, I initially wanted to update the base packages on CentOS 7 before proceeding. However, given the circumstances, it seems deploying a separate AlmaLinux machine for testing might be the most efficient approach to troubleshooting the issue. If the problem persists on a fresh AlmaLinux

We've got our Polycom phones auto-answering for paging. Is it possible to configure a PAP2 to auto-answer for either paging or intercom? If so, how?

Hello, all. We have successfully connected our new Asterisk 1.6.1.1 PBX to Vitelity's network and have been very happy with them thus far. However, we'd like to use domains in our sip.conf to facilitate routing in our multi-tenant environment. We also like to set allowexternaldomains=no for security. However, this breaks our inbound PSTN calling from Vitelity. Is it possible to use

On 16 Nov 2001, Zaheer Merali wrote: > That is an idea that is coming up in the next ZStreamCaster. > ZStreamcaster 0.1 currently allows you to save a stream to disk at a > higher bitrate than you send to the icecast server at. > > I am planning to add a feature that allows you to have n streams going > out, each for different bitrates (or alternatively different sample >

Greetings, I've noticed a problem that might originate from my Asterisk configuration, could use a hand in sorting it out. Problem is a 488 response from Asterisk whenever it gets RTP/SAVPF profile in the SDP. My current setup has Asterisk Kamailio realtime integration, and Kamailio uses dispatcher to route calls for Asterisk to handle. Now I have only one Asterisk, on the same machine as

Hi All, I'm having a bit of trouble with my new *NOW setup. I've downloaded and installed *NOW 1.5. We're using 1 SIP Trunk from SimpleSignal.com. Outbound calling works great, but I'm having some trouble with inbound calls. First, we would get the "the number you have dialed is not in service" error on inbound calls. After some googling, I found out that I needed